We use FreePBX to, amongst other things, connect users to audio couplers over POTS.
This process works great when a call from a softphone (e.g. PhonerLite) is actually torn down properly with an explicit hangup notification, but when network connections drop, we can’t seem to get the RTP stream on the trunk side to either time out or otherwise hang up when the user-side SIP peer has dropped unexpectedly.
Put another way, the connection looks like this:
audio source <—> Audio coupler <—A1—> POTS <—A2—> SIP Trunk <—B2—> FreePBX <—B1—> Softphone
We want the call to drop/timeout in the A1 to A2 part of the chain (not the Trunk registration at B2, obviously) to disconnect when the call at B1 does not end gracefully.
I’ve played with the RTP timeout in SIP settings, but i suspect that since there’s still audio / RTP traffic coming from the A1 endpoint, FreePBX doesn’t know to timeout.
Hoping that someone has seen this setup or something like it before and can at least point me in the right direction.
Happy Holidays, all!